Administrator’s Guide SoundPoint IP / SoundStation IP

 

until the registration is successful (for example, when the Internet link is once

 

again operational). While the primary server registration is unavailable, the

 

next highest priority server in the list will serve as the working server. As soon

 

as the primary server registration succeeds, it will return to being the working

 

server.

Note

 

If reg.x.server.y.register is set to 0, then phone will not register to that server.

 

However, the INVITE will fail over to that server if all higher priority servers are

 

down.

 

 

 

Recommended Practices for Fallback Deployments

 

In situations where server redundancy for fall-back purpose is used, the

 

following measures should be taken to optimize the effectiveness of the

 

solution:

 

1. Deploy an on-site DNS server to avoid long call initiation delays that can

 

result if the DNS server records expire.

 

2. Do not use OutBoundProxy configurations on the phone if the

 

OutBoundProxy could be unreachable when the fallback occurs.

 

SoundPoint IP phones can only be configured with one OutBoundProxy

 

per registration and all traffic for that registration will be routed through

 

this proxy for all servers attached to that registration. If Server 2 is not

 

accessible through the configured proxy, call signaling with Server 2 will

 

fail.

 

3. Avoid using too many servers as part of the redundancy configuration as

 

each registration will generate more traffic.

 

4. Educate users as to the features that will not be available when in

 

“fallback” operating mode.

Presence

The Presence feature allows the phone to monitor the status of other users/devices and allows other users to monitor it. The status of monitored users is displayed visually and is updated in real time in the Buddies display screen or, for speed dial entries, on the phone’s idle display. Users can block others from monitoring their phones and are notified when a change in monitored status occurs. Phone status changes are broadcast automatically to monitoring phones when the user engages in calls or invokes do-not-disturb. The user can also manually specify a state to convey, overriding, and perhaps masking, the automatic behavior.

Note

Notification when a change in monitored status occurs will be available in a

 

subsequent release.

 

 

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Polycom SIP 3.1 manual Presence
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SIP 3.1 specifications

Polycom SIP 3.1 is an advanced session initiation protocol designed to enhance voice and video communication in various business environments. As a pivotal component of Polycom’s telecommunication solutions, SIP 3.1 offers several features and characteristics that cater to the evolving needs of modern enterprises, particularly those that rely on seamless and efficient communication.

One of the standout features of Polycom SIP 3.1 is its robust interoperability. This protocol supports a wide range of endpoints and platforms, allowing organizations to integrate their existing systems with new technologies effortlessly. This flexibility ensures that businesses can leverage their previous investments while upgrading to the latest communication tools.

Another key aspect of Polycom SIP 3.1 is its enhanced call management capabilities. The protocol facilitates efficient call handling, enabling users to manage multiple calls seamlessly. Features like call hold, transfer, and conferencing are streamlined, which enhances productivity by allowing for efficient collaboration among team members. Additionally, it is optimized for low latency and high-quality audio, ensuring that conversations are clear and free from disruptions.

Security is paramount in today’s digital landscape, and Polycom SIP 3.1 addresses this concern with advanced encryption standards. By integrating security features such as Transport Layer Security (TLS) and Secure Real-time Transport Protocol (SRTP), it protects sensitive communication from unauthorized access and ensures that data remains confidential throughout the call.

Polycom SIP 3.1 also boasts compatibility with various video codecs, making it a versatile choice for video conferencing. This compatibility ensures high-quality video streams, which is essential for effective visual communication in remote meetings. Furthermore, the support for the H.264 codec provides efficient bandwidth usage, making high-definition video conferencing accessible, even in varying network conditions.

Moreover, the protocol provides strong support for presence and instant messaging, which enhances real-time communication among users. This integration of voice, video, and messaging capabilities fosters a more connected and collaborative work environment, allowing teams to engage effectively regardless of their geographical locations.

In summary, Polycom SIP 3.1 stands out as a sophisticated solution tailored to meet the demands of modern business communication. With its emphasis on interoperability, call management, security, video quality, and real-time collaboration, it caters to companies of all sizes seeking to optimize their communication infrastructure and enhance productivity in the workplace. As businesses continue to navigate the complexities of digital communication, Polycom SIP 3.1 remains a compelling choice in the market.
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